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Braindumps for "1Y0-973" Exam

HIP!HIP! HURRAY.....PASSED WITH 94%

 I PASSED WITH 94%, ITCERTKEYS IS THE GREAT HELP FOR ME..

I APLOAD SOME QUESTIONS FROM ITCERTKEYS STUDY GUIDE.

Question 1.
You want to ensure that the agent prompts users to reauthenticate after a specified length of time. Which setting in the Access manager node can be used to perform this task?

A. Auto Logo in
B. Allow Reveal
C. Allow Refresh
D. Logon After Config

Answer: A

Question 2.
You have installed the Meta frame Password Manager agent for all users in the Sales department. All Sales applications share on Oracle database for user authentication. The department has requested that Meta frame Password Manager automatically adjust all the Sales Applications credentials whenever the password changes for one of them. Password sharing has already been enabled.

What two actions need to be taken? (Choose two.)

A. Create a new password policy group called Sales.
B. Crete a new password sharing group called Sales.
C. Configure each Sales application to use the Sales password policy.
D. Configure each Sales application to use the Sales password sharing group.
E. Configure each Sales application to use the Domain password sharing group.

Answer: B, D

Question 3.
Which statement is true about a default agent installation?

A. Users can add new three-field web applications.
B. Users can add new two-field Windows applications.
C. Users can delete pre-defined applications from the Logon Manager.
D. Users can add applications that require user ID, password, and domain.

Answer: B

Question 4.
You are the Meta frame Password Manager administrator for ITCertKeys.com. Mary, the Help Desk manager, has contacted you about troubleshooting issues for Meta frame Password Manager. She would like to find out if something can be done to allow the help desk personnel to easily identify which Meta frame Presentation Server is generating the Meta frame Password Manager session for users.

Which actions satisfies Mary's request?

A. In Agent Settings -> Shell, set the Auto Logon Delay setting to 3
B. In Agent Settings -> Shell, set the Display Computer Name setting to Enabled
C. IN Agent Settings -> Access Manager, set the Allow Reveal setting to Enabled
D. In Agent Settings -> Shell, set the Show Tray Icon setting to "Show The Tray Icon."

Answer: B

Question 5.
An administrator will implement Meta frame Password Manager to provide SSO for users who access published applications using Web Interface for Meta frame Presentation Server. Users currently access applications through a browser using an ICA client. Remote connections are secured through the use of Secure Gateway for Meta frame Presentation Server.

In which location should the agent be installed for this deployment scenario?

A. On the server running the Secure Gateway service
B. On each Meta frame Presentation Server in the farm
C. On the system with the Secure Ticketing Authority installed
D. On the Internet Information Server hosting Web Interface for Meta frame Presentation Server

Answer: B

Question 6.
When configuring an application it is important that the application is _______. (Fill in the blank with the correct option.)

A. published on a server running Meta frame Presentation Server
B. installed in the same place as the Meta frame Password Manager agent
C. listed in the application list of the Meta frame Password Manager console
D. running on the same operating system as the Meta frame Password Manager agent

Answer: D



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Braindumps: Dumps for HP0-K02 Exam Brain Dump

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Braindumps for "HP0-K02" Exam

Implementing HP BladeSystem

 Question 1.
Which function does the Onboard Administrator perform in the BladeSystem c-Class? (Select two.)

A. configures and troubleshoots the server blade
B. manages connections to the interconnect modules
C. activates Remote Deployment Pack (RDP) when an in-service server fails
D. automates the process of deploying server operating systems and software
E. drives all management features through the Inter-Integrated Circuit (I2C) and Intelligent
Chassis Management Bus (ICMB) interfaces

Answer: B, E

Question 2.
Which options properly power down a c-Class blade server? (Select two.)

A. Use the virtual power button selection through iLO2.
B. Remove the c-Class blade server from the enclosure.
C. Use the virtual power button selection through RILOE II.
D. Switch off the power supplies on the back of the c7000 enclosure.
E. Press and hold the Power On/Standby button for more than 4 seconds.

Answer: A, E

Question 3.
What happens if you install extra fans in bays of a c7000 BladeSystem enclosure in such a way that they do not meet the next higher fan rule?

A. The enclosure will shut down.
B. The new fans will spin up and run at full speed.
C. The red LED will start blinking on the new fans.
D. The new fans will be marked with location errors.

Answer: D

Question 4.
Which LED indicator shows a fan failure?

A. solid blue
B. solid red
C. solid green
D. solid amber

Answer: D

Question 5.
What is the correct way of removing an active cool fan from a c-Class enclosure?

A. Lift the handle upwards and pull the fan from the enclosure.
B. Turn the handle on the fan counterclockwise and remove the fan.
C. Remove the four screws securing the fan and lift it out of the enclosure.
D. Remove the power connector from the fan and lift the fan out of the chassis.

Answer: B

Question 6.
Which port on a BladeSystem c-Class enclosure serves as a service port for temporary connection to a PC?

A. USB port
B. serial port
C. enclosure link-up port
D. enclosure link-down port

Answer: C

Question 7.
Exhibit:
 

You want to enable direct FC SAN connectivity on a c7000 enclosure. 

Where must you install the primary fabric switch?

A. Position 6
B. Position 18
C. Position 21
D. Position 22

Answer: C

Question 8.
Which features are available in Instant Support Enterprise Edition (ISEE) Advanced? (Select three.)

A. monthly OS update and patch downloads
B. real-time hardware event monitoring and automated notification to the HP 3rd level support
C. event management and fault detection to enhance Support Point of Presence (SPOP) 
    availability
D. comprehensive enterprise view of open incidents, history, installed clients, client status, and 
    contact information
E. convenient central point of administration through an HP owned and maintained Support Point 
    of Presence (SPOP) server installed at the customer site

Answer: C, D, E

Question 9.
Which HP service offering provides a two-hour response time for software issues and a four-hour response time for hardware issue resolution?

A. Proactive 24
B. HP Support Plus
C. HP Support Plus 24
D. Software Plus Support Service

Answer: A

Question 10.
What should a customer do before calling for service during the warranty period?

A. Install the ISEE software to automatically log the event.
B. Order a Care Pack online to have service on the same day.
C. Download the maintenance and service guide for the server.
D. Run diagnostics and install the latest firmware, drivers and patches.

Answer: D



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Braindumps: Dumps for 646-204 Exam Brain Dump

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Braindumps for "646-204" Exam

Cisco Sales Expert

 Question 1.
Which two characteristics of optical networks data to be transmitted over extremely long distances? (Choose two.)

A. No electromagnetic interference (EMI)
B. Minimal signal loss
C. No bandwidth limits
D. No encryption

Answer: A, B

Question 2.
What is a wide-area network (WAN)?

A. A network that is restricted to a single building, a group of buildings, or even a single room
B. A network that connects client devices wirelessly using radio waves
C. A network that crosses metropolitan, regional or national boundaries
D. A network that encompasses an entire city or metropolitan area and connects multiple local-
    area networks (LANs)

Answer: C

Question 3.
Which IP telephony component provides call processing, such as signaling that in a traditional voice network is provided by switching equipment, such as a PBX?

A. Voice gateways, such as the Cisco VG248 analog phone gateway
B. Call-management hardware and software, such as Cisco Call Manager
C. Client-based software, such as Cisco SoftPhone
D. Advanced call-handling software, such as Cisco Unity

Answer: B

Question 4.
A software engineering firm wants to streamline call center operations to reduce customer wait times and improve employee productivity. 

Which Cisco solution best matches this business need?

A. IP Telephony
B. DSL
C. WLAN
D. Optical Network

Answer: A

Question 5.
A growing publishing firm needs a secure way for remote content editors to connect through the Internet to the company extranet.

Which Cisco solution best matches this business need?

A. IP telephony
B. Virtual Private Network (VPN)
C. Content Networking
D. Wireless network

Answer: B

Question 6.
What creates an encrypted tunnel through a shared public network infrastructure, such as the Internet to provide a secure connection between remote users and a private network?

A. VPN
B. HTTP
C. DSL
D. Optical Networking

Answer: A

Question 7.
What does a router do?

A. Reads Media Access Control (MAC) addresses to forward messages to the correct location 
    and connects the local-area network (LAN) to the wide-area network (WAN)
B. Stores data on network and connects the local-area network (LAN) to the wide-area network 
    (WAN)
C. Routes traffic down alternative paths and connects the local-area network (LAN) to the wide-
    area network (WAN)
D. Broadcasts data to all devices that are connected to it across the local-area network (LAN) and 
    wide-area network (WAN)

Answer: C

Question 8.
Which device is used in wireless networks to send transmission to a target area?

A. Directional Antenna
B. Optical fiber
C. Omni-directional antenna
D. Access point

Answer: A

Question 9.
Which wireless local-area network (WLAN) device is often used to make wireless connections between buildings or campuses?

A. Access Point
B. Bridge
C. Router
D. Client Adapter

Answer: B

Question 10.
On what type of network does the Cisco Unified Communications system reside?

A. A single, joined voice, video and data network
B. Voice networks only
C. Wireless networks only
D. Data networks only
E. Independent voice, video and data networks

Answer: A


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Braindumps: Dumps for 642-845 Exam Brain Dump

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Braindumps for "642-845" Exam

Optimizing Converged Cisco Networks

 Question 1.
You need to implement QoS for the ITCertKeys VOIP network. 

Which three statements are true about the data traffic characteristics of voice traffic? (Select three)

A. Voice packets require TCP for rapid retransmission of dropped packets.
B. Latency is not a concern as long as jitter is kept below 30 ms.
C. Voice packets require a fairly constant bandwidth reserved for voice control traffic as well as 
    for the voice payload.
D. Voice packets do not require a specific type of queuing.
E. Latency must be kept below 150 ms.
F. Voice packets are rather small

Answer: C, E, F

Explanation:
QoS refers to the ability of a network to provide improved service to selected network traffic over various underlying technologies including Frame Relay, ATM, Ethernet and 802.3 networks, SONET, and IP-routed networks.

QoS features provide improved and more predictable network service by offering the following services:
1. Dedicated bandwidth
2. Improved loss characteristics
3. Congestion management and avoidance
4. Traffic shaping
5. Prioritization of traffic
Voice quality is directly affected by all three QoS quality factors such as loss, delay, and delay variation.

Loss causes voice clipping and skips. Industry standard codec algorithms can correct for up to 30 ms of lost voice. Cisco Voice over IP (VoIP) technology uses 20 ms samples of voice payload per VoIP packet. Only a single Real Time Transport (RTP) packet could be lost at any given time. If two successive voice packets are lost, the 30 ms correctable window is exceeded and voice quality begins to degrade. Delay can cause voice quality degradation if it is above 200 ms. If the end-to-end voice delay becomes too long, the conversation sounds as if two parties are talking over a satellite link or a CB radio. The ITU standard for VoIP, G.114, states that a 150 ms one-way delay budget is acceptable for high voice quality. With respect to delay variation, there are adaptive jitter buffers within IP Telephony devices. These buffers can usually compensate for 20 to 50 ms of jitter.

Question 2.
ITCertKeys uses G.711 for the VOIP calls. When analog signals are digitized using the G.711 codec, voice samples are encapsulated into protocol data units (PDUs) involving which three headers? (Select three)

A. UDP
B. RTP
C. IP
D. TCP
E. Compressed RTP
F. H.323

Answer: A, B, C

Explanation:
When a VoIP device, such as a gateway, sends voice over an IP network, the digitized voice has to be encapsulated into an IP packet. Voice transmission requires features not provided by the IP protocol header; therefore, additional transport protocols have to be used. Transport protocols that include features required for voice transmission are TCP, UDP, and RTP. VoIP utilizes a combination of UDP and RTP.

Question 3.
VOIP has been rolled out to every ITCertKeys location. 

What are three features and functions of voice (VOIP) traffic on a network? (Select three)

A. Voice traffic is bursty
B. Voice traffic is retransmittable
C. Voice traffic is time-sensitive
D. Voice traffic is bandwidth intensive
E. Voice traffic is constant
F. Voice traffic uses small packet sizes

Answer: C, E, F

Explanation:
The benefits of packet telephony networks include 
i. More efficient use of bandwidth and equipment: Traditional telephony networks use a 64-kbps channel for every voice call. Packet telephony shares bandwidth among multiple logical connections.
ii. Lower transmission costs: A substantial amount of equipment is needed to combine 64-kbps channels into high-speed links for transport across the network. Packet telephony statistically multiplexes voice traffic alongside data traffic. This consolidation provides substantial savings on capital equipment and operations costs.
iii. Consolidated network expenses: Instead of operating separate networks for voice and data, voice networks are converted to use the packet-switched architecture to create a single integrated communications network with a common switching and transmission system. The benefit is significant cost savings on network equipment and operations.
iv. Improved employee productivity through features provided by IP telephony: IP phones are not only phones, they are complete business communication devices. They offer directory lookups and access to databases through Extensible Markup Language (XML) applications. These applications allow simple integration of telephony into any business application. For instance, employees can use the phone to look up information about a customer who called in, search for inventory information, and enter orders. The employee can be notified of a issue (for example, a change of the shipment date), and with a single click can call the customer about the change. In addition, software-based phones or wireless phones offer mobility to the phone user.

Question 4.
ITCertKeys is rolling out an H.323 VOIP network using Cisco devices. 

Which IOS feature provides dial plan scalability and bandwidth management for H.323 VoIP implementations?

A. Digital Signal Processors
B. Call Routing
C. Gatekeeper
D. Call Admission Control
E. None of the above

Answer: C

Explanation:
Enterprise voice implementations use components such as gateways, gatekeepers, Cisco Unified CallManager, and IP phones. Cisco Unified CallManager offers PBX-like features to IP phones. Gateways interconnect traditional telephony systems, such as analog or digital phones, PBXs, or the public switched telephone network (PSTN) to the IP telephony solution. Gatekeepers can be used for scalability of dial plans and for bandwidth management when using the H.323 protocol.

Question 5.
A Cisco router is being used as a VOIP gateway to convert voice signals in the ITCertKeys network. 

What steps are taken when a router converts a voice signal from analog to digital form? (Select two)

A. Quantization
B. Serialization
C. Packetization
D. Sampling

Answer: A, D

Explanation:
Step 1 Sampling: The analog signal is sampled periodically. The output of the sampling is a pulse amplitude modulation (PAM) signal. 
Step 2 Quantization: The PAM signal is matched to a segmented scale. This scale measures the amplitude (height) of the PAM signal.
Step 3 Encoding: The matched scale value is represented in binary format.
Step 4 Compression: Optionally, voice samples can be compressed to reduce bandwidth requirements. Analog-to-digital conversion is done by digital signal processors (DSPs), which are located on the voice interface cards. The conversion is needed for calls received on analog lines, which are then sent out to a packet network or to a digital voice interface.

Question 6.
You need to implement the proper IOS tools to ensure that VOIP works over the ITCertKeys network. 

Which queuing and compression mechanisms are needed to effectively use the available bandwidth for voice traffic? (Select two)

A. Priority Queuing (PQ) or Custom Queuing (CQ)
B. Real-Time Transport Protocol (RTP) header compression
C. Low Latency Queuing (LLQ)
D. Class-Based Weighted Fair Queuing (CBWFQ)
E. TCP header compression
F. UDP header compression

Answer: D, E

Explanation:
1. Class-based weighted fair queuing (CBWFQ) extends the standard WFQ functionality to provide support for user-defined traffic classes. By using CBWFQ, network managers can define traffic classes based on several match criteria, including protocols, access control lists (ACLs), and input interfaces. A FIFO queue is reserved for each class, and traffic belonging to a class is directed to the queue for that class. More than one IP flow, or "conversation", can belong to a class. Once a class has been defined according to its match criteria, the characteristics can be assigned to the class. To characterize a class, assign the bandwidth and maximum packet limit. The bandwidth assigned to a class is the guaranteed bandwidth given to the class during congestion.

CBWFQ assigns a weight to each configured class instead of each flow. This weight is proportional to the bandwidth configured for each class. Weight is equal to the interface bandwidth divided by the class bandwidth. Therefore, a class with a higher bandwidth value will have a lower weight. By default, the total amount of bandwidth allocated for all classes must not exceed 75 percent of the available bandwidth on the interface. The other 25 percent is used for control and routing traffic. The queue limit must also be specified for the class. The specification is the maximum number of packets allowed to accumulate in the queue for the class. Packets belonging to a class are subject to the bandwidth and queue limits that are configured for the class. 

2. TCP/IP header compression subscribes to the Van Jacobson Algorithm defined in RFC 1144. TCP/IP header compression lowers the overhead generated by the disproportionately large TCP/IP headers as they are transmitted across the WAN. TCP/IP header compression is protocol-specific and only compresses the TCP/IP header. The Layer 2 header is still intact and a packet with a compressed TCP/IP header can still travel across a WAN link. TCP/IP header compression is beneficial on small packets with few bytes of data such as Telnet. Cisco's header compression supports Frame Relay and dial-on-demand WAN link protocols. Because of processing overhead, header compression is generally used at lower speeds, such as 64 kbps links.

Question 7.
You want to ensure the highest call quality possible for all VOIP calls in the ITCertKeys network. 

Which codec standard would provide the highest voice-quality, mean opinion score (MOS)?

A. G.711, PCM
B. G.729, CS-ACELP
C. G.729A, CS-ACELP
D. G.728, LDCELP
E. None of the above

Answer: A

Explanation:
When a call is placed between two phones, the call setup stage occurs first. As a result of this process, the call is logically set up, but no dedicated circuits (lines) are associated with the call. The gateway then converts the received analog signals into digital format using a codec, such as G.711 or G.729 if voice compression is being used. When analog signals are digitized using the G.711 codec, 20 ms of voice consists of 160 samples, 8 bits each. The result is 160 bytes of voice information. These G.711 samples (160 bytes) are encapsulated into an RTP header (12 bytes), a UDP header (8 bytes), and an IP header (20 bytes). Therefore, the whole IP packet carrying UDP, RTP, and the voice payload has a size of 200 bytes. When G.711 is being used, the ratio of header to payload is smaller because of the larger voice payload. Forty bytes of headers are added to 160 bytes of payload, so one-fourth of the G.711 codec bandwidth (64 kbps) has to be added. Without Layer 2 overhead, a G.711 call requires 80 kbps.

Question 8.
When a router converts analog signals to digital signals as part of the VoIP process, it performs four separate steps. 

From the options shown below, which set of steps contains the steps in their correct sequence?

A. encoding
    quantization
    optional compression
    sampling
B. optional compression
    encoding
    sampling
    quantization
C. sampling
    quantization
    encoding
    optional compression
D. optional compression
    sampling
    encoding
    quantization
E. sampling
    quantization
    optional compression
    encoding
F. encoding
    optional compression
    quantization
    sampling
G. None of the above

Answer: C

Explanation:
Step 1 Sampling: The analog signal is sampled periodically. The output of the sampling is a pulse amplitude modulation (PAM) signal.
Step 2 Quantization: The PAM signal is matched to a segmented scale. This scale measures the amplitude (height) of the PAM signal.
Step 3 Encoding: The matched scale value is represented in binary format.
Step 4 Compression:
Optionally, voice samples can be compressed to reduce bandwidth requirements. Analog-to-digital conversion is done by digital signal processors (DSPs), which are located on the voice interface cards. The conversion is needed for calls received on analog lines, which are then sent out to a packet network or to a digital voice interface. 

Question 9.
ITCertKeys has determined that during its busiest hours, the average number of internal VoIP calls across the WAN link is four (4). Since this is an average, the WAN link has been sized for six (6) calls with no call admission control. 

What will happen when a seventh call is attempted across the WAN link?

A. The seventh call is routed via the PSTN.
B. The call is completed, but all calls have quality issues.
C. The call is completed but the seventh call has quality issues.
D. The call is denied and the original six (6) calls remain.
E. The call is completed and the first call is dropped.
F. None of the above.

Answer: B

Explanation:
IP telephony solutions offer Call Admission Control (CAC), a feature that artificially limits the number of concurrent voice calls to prevent oversubscription of WAN resources. Without CAC, if too many calls are active and too much voice traffic is sent, delays and packet drops occur. Even giving Real-Time Transport Protocol (RTP) packets absolute priority over all other traffic does not help when the physical bandwidth is not sufficient to carry all voice packets. Quality of service (QoS) mechanisms do not associate individual RTP packets with individual calls; therefore, all RTP packets are treated equally. All RTP packets will experience delays, and any RTP packets may be dropped. The effect of this behavior is that all voice calls experience voice quality degradation when oversubscription occurs. It is a common misconception that only calls that are beyond the bandwidth limit will suffer from quality degradation. CAC is the only method that prevents general voice quality degradation caused by too many concurrent active calls.

Question 10.
While planning the new ITCertKeys VOIP network, you need to determine the size of the WAN links to use. To do this, you need to calculate the bandwidth required by each call. 

Which three pieces of information are used to calculate the total bandwidth of a VoIP call? (Select three)

A. The serialization of the interface
B. The quantization
C. The TCP overhead
D. The packetization size
E. The UDP overhead
F. The packet rate

Answer: D, E, F

Explanation:
Packet rate: Packet rate specifies the number of packets sent in a certain time interval. The packet rate is usually specified in packets per second (pps). Packet rate is the multiplicative inverse of the packetization period. The packetization period is the amount of voice (time) that will be encapsulated per packet, and is usually specified in milliseconds.
Packetization size: Packetization size specifies the number of bytes that are needed to represent the voice information that will be encapsulated per packet. Packetization size depends on the packetization period and the bandwidth of the codec used. 
IP overhead: IP overhead specifies the number of bytes added to the voice information during IP encapsulation. When voice is encapsulated into Real-Time Transport Protocol (RTP), User Datagram Protocol (UDP), and IP, the IP overhead is the sum of all these headers.
Data link overhead: Data-link overhead specifies the number of bytes added during data-link encapsulation. The data-link overhead depends on the used data-link protocol, which can be different per link.
Tunneling overhead: Tunneling overhead specifies the number of bytes added by any security or tunneling protocol, such as 802.1Q tunneling, IPsec, Generic Route Encapsulation (GRE), or Multiprotocol Label Switching (MPLS). This overhead must be considered on all links between the tunnel source and the tunnel destination.



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Braindumps: Dumps for 640-802 Exam Brain Dump

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Braindumps for "640-802" Exam

Cisco Certified Network Associate

 Question 1.
Both switches and hubs are being utilized within the ITCertKeys network. 

Which of the following is true regarding the use of switches and hubs for network connectivity in this network?

A. Switches take less time to process frames than hubs take
B. Hubs can filter frames
C. Switches do not forward broadcasts
D. Switches increase the number of collision domains in the network
E. Using hubs can increase the amount of bandwidth available to hosts
F. None of the above

Answer:  D

Explanation:
The biggest benefit of using switches instead of hubs in your internetwork is that each switch port is actually its own collision domain. (Conversely, a hub creates one large collision domain.) But even armed with a switch, you still can't break up broadcast domains. Neither switches nor bridges will do that. They'll typically simply forward all broadcasts instead. Switch creates the collision domain per port, so we can say switch increase the number of collision domains.

Question 2.
Which one of the following characteristics is true regarding the use of hubs and switches?

A. Hubs can have their ports be configured with VLANs
B. Using hubs is costly with regard to bandwidth availability.
C. Switches can not forward broadcasts.
D. Switches are more efficient than hubs in processing frames.
E. Switches increase the number of collision domains in the network.

Answer:  E

Explanation: 
Switches increases the number of collisions domains in the network. Switches that are configured with VLANs will reduce the size of the collision domains by increasing the number of collision domains in a network, but making them smaller than that of one big, flat network.

Incorrect Answers:
A. Switches are capable of VLAN configurations, but hubs are not.
B. Hubs are generally the least costly method possible to connect multiple devices together in a 
    network.
C. Switches forward broadcasts and multicasts, by default, to all ports within the same VLAN. 
    Only routers block all broadcast traffic by default.
D. Switches and hubs can be equally efficient in processing frames, in theory. In practice, 
    switches are generally more efficient as they usually have more CPU and memory allocated to 
    them, and are generally much more expensive than a simple hub.

Question 3.
When comparing and contrasting the similarities and differences between bridges and switches, which of the following are valid statements? Choose all the valid answer choices)

A. Bridges are faster than switches because they have fewer ports.
B. A switch is a multiport bridge,
C. Bridges and switches learn MAC addresses by examining the source MAC address of each 
    frame received.
D. A bridge will forward a broadcast but a switch will not.
E. Bridges and switches increase the size of a collision domain.
F. None of the above statements are true.

Answer:  B, C

Explanation:
Both bridges and switches build the bridge table by listening to incoming frames and examining the source MAC address in the frame. Switches are multiport bridges that allow you to create multiple broadcast domains. Each broadcast domain is like a distinct virtual bridge within a switch.

Incorrect Answers:
A. Switches are generally faster than bridges. Bridges also do not necessarily have fewer ports 
    than switches.
D. Both bridges and switches will forward broadcast and multicast traffic, assuming that the traffic 
    remains in the same VLAN.
E. The use of VLANs in a switch can decrease the size of the collision domain, by creating 
    additional, smaller collision domains.

Question 4.
Which of the following correctly describe the various functions and virtues of a router? (Select all valid answer choices)

A. Packet switching
B. Collision prevention on a LAN segment.
C. Packet filtering
D. Broadcast domain enlargement
E. Broadcast forwarding
F. Internetwork communication
G. None of the above

Answer:  A, C, F

Explanation:
The main function of a router is to connect different, separated networks together. In doing so, switching packets from one network to another is a primary function, along with providing for communication between networks. As an additional feature, routers are capable of providing filtering on a network address and application port level, so choice C is also correct.

Incorrect Answers:
B. Routers can indeed be used to segment a network separate a collision domain, since routers 
    do not forward LAN broadcasts and multicasts to other interfaces. However, routers alone can 
    not prevent all collisions from occurring on any given LAN segment.
D. Routers actually segment LANs into smaller broadcast domains.
E. Routers do not forward broadcast and multicast traffic out the additional interfaces by default. 
    Unless bridging or IP helpers are configured on the router, LAN broadcasts are blocked at the 
    router level.

Question 5.
The LAN needs are expanding at the ITCertKeysCorporate office, which is quickly growing. You are instructed to enlarge the area covered by a single LAN segment on the ITCertKeys network.

Which of the following are layer 1 devices that you can use? (Choose all that apply.)

A. A switch
B. A router
C. A network adapter card
D. A hub
E. A repeater

Answer:  D, E

Explanation:
A hub simply repeats the electrical signal and makes no attempt to interpret the electrical signal (layer 1) as a LAN frame (Layer 2). So, a hub actually performs OSI layer 1 functions, repeating an electrical signal, whereas a switch performs OSI layer 2 functions, actually interpreting Ethernet header information, particularly addresses, to make forwarding decisions. Hubs can be used to increase the number of stations that can be supported on a LAN. Because the repeater does not interpret what the bits mean, but does examine and generate electrical signals, a repeater is considered to operate at Layer 1. Repeaters can be used to physically extend the LAN to greater distances.

Question 6.
Cisco is the leader in the router market space. 

What basic functions do their routers perform in a network? (Choose two)

A. The microsegmentation of broadcast domains
B. Path selection
C. Packet switching
D. Bridging between LAN segments
E. Access layer security
F. VLAN membership assignment
G. Application optimization

Answer:  B, C

Explanation:
The primary functions of a router are: Packet Switching and Path Selection. It is the routers job to determine the best method for delivering the data, and switching that data as quickly as possible.

Question 7.
Both bridges are switches are being used throughout the ITCertKeys LAN. 

Which of the following statements are true regarding bridges and switches in this network? (Choose 3)

A. Switches are primarily software based while bridges are hardware based.
B. Switches usually have a higher number of ports than most bridges.
C. Bridges are frequently faster than switches.
D. Bridges define broadcast domains while switches define collision domains.
E. Both bridges and switches forward Layer 2 broadcasts.
F. Both bridges and switches make forwarding decisions based on Layer 2 addresses.

Answer:  B, E, F

Question 8.
As a network administrator, you will need to decide on the appropriate network devices to use. 

Which of the following correctly describes the roles of devices in a WAN? (Choose three)

A. A CSU/DSU terminates a digital local loop.
B. A router is commonly considered a DCE device.
C. A modem terminates an analog local loop.
D. A router is commonly considered a DTE device.
E. A modem terminates a digital local loop.
F. A CSU/DSU terminates an analog local loop.
G. A modem is used to terminate a T1

Answer:  A, C, D

Explanation:
Layer 2 switching is considered hardware-based bridging because it uses specialized hardware called an application-specific integrated circuit (ASIC). ASICs can run up to gigabit speeds with very low latency rates. Switches usually have higher port number then bridge. Generally bridges have two ports. Both operates on Data link layer.

Question 9.
The ITCertKeys network administrator needs to determine what LAN devices to install on the ITCertKeys network. 

What are two advantages of using Layer 2 Ethernet switches over hubs? (Choose two)

A. Allowing simultaneous frame transmissions
B. Increasing the size of broadcast domains
C. Increasing the maximum length of UTP cabling between devices
D. Filtering frames based on MAC addresses
E. Decreasing the number of collision domains

Answer:  A, D

Explanation:
A: A half duplex connection is where only one device can send or receive at a time. A full duplex connection is where both devices can send and receive at the same time. Thus, if you have a 100Mb half-duplex connection, only sending at 100Mb OR receiving at 100Mb can happen at the same time. If you have a 100Mb full duplex connection, you can effectively get 200Mb out of the link because you could be sending 100Mb and receiving 100Mb at the same time.
D: Switches are capable of filtering frames based on any Layer 2 fields. For example, a switch can be programmed to reject (not forward) all frames sourced from a particular network. Because link layer information often includes a reference to an upper-layer protocol, switches usually can filter on this parameter. Furthermore, filters can be helpful in dealing with unnecessary broadcast and multicast packets.

Question 10.
CDP is being used throughout the ITCertKeys network. 

What are two reasons why the ITCertKeys network administrator would use CDP? (Choose two)

A. To determine the status of network services on a remote device
B. To obtain the IP Address of a connected device in order to telnet to the device
C. To verify the type of cable interconnecting two devices
D. To verify Layer 2 connectivity between two devices when Layer 3 fails
E. To obtain VLAN information from directly connected switches
F. To determine the status of the routing protocols between directly connected routers
G. To support automatic network failover during outages

Answer:  B, D

Explanation:
Cisco Discovery Protocol (CDP) is a proprietary protocol designed by Cisco to help administrators collect information about both locally attached and remote devices. By using CDP, you can gather hardware and protocol information about neighbor devices, which is useful info for troubleshooting and documenting the network. 
You can use:
Show cdp neighbor
Show cdp neighbor details
Commands to gather the information of connected neighbors.


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Braindumps for "642-873" Exam

Designing Cisco Network Service Architectures Exam

 Question 1.
DRAG DROP
You work as a network technician at ITCertKeys.com. Your boss, Mrs. ITCertKeys, in Cisco client security software applications. Match the applications with the descriptions.
 
Answer:

Question 2.
Which two statements about Layer 3 access designs are correct? (Choose two.)

A. Broadcast and fault domains are increased
B. Limits on clustering and NIC teaming are removed
C. Convergence time is fractionally slower than STP
D. IP Address space is difficult to manage
E. Fast uplink convergence is supported for failover and fallback

Answer: D, E

Question 3.
Which two of these key fields are used to identify a flow in a traditional NetFlow implementation? (Choose two.)

A. Destination IP Address
B. Output Interface
C. Source MAC address
D. Next-hop IP Address
E. Source Port
F. Next-hop MAC Address

Answer: A, E

Question 4.
Which of these is true of IP addressing with regard to VPN termination?

A. Termination devices need routable addresses inside the VPN
B. Designs should not include overlapping address spaces between sites, since NAT is not supported
C. IGP routing protocols will update their routing tables over an IPSec VPN
D. Addressing designs need to allow for summarization

Answer: D

Question 5.
When is a first-hop redundancy protocol needed in the distribution layer?

A. When preempt tuning of the default gateway is needed
B. When the design implements Layer 2 between the access switch and the distribution switch
C. When HSRP is not supported by the design
D. When a robust method of backing up the default gateway is needed
E. When multiple vendor devices need to be supported
F. When the design implements Layer 3 between the access switch and the distribution switch

Answer: F

Question 6.
Which of these practices should you follow when designing a Layer 3 routing protocol?

A. Never peer on transit links
B. Build squares for deterministic convergence
C. Build inverted U designs for deterministic convergence
D. Summarize routes at the distribution to the core to limit EIGRP queries or OSPF LSA propagation

Answer: D

Question 7.
Which two statements correctly identify considerations to take into account when deciding on Campus QoS Design elements? (Choose two.)

A. Call signaling must have guaranteed bandwidth service
B. Strict-priority queuing should be limited to 50% of the capacity of the link
C. At least 23 percent of the link bandwidth should be reserved for default best-effort class
D. Voice needs to be assigned to the hardware priority queue
E. Voice needs to be assigned to the software priority queue

Answer: A, D

Question 8.
Which statement about IDS/IPS design is correct?

A. An IPS should be deployed if the security policy does not support the denial of traffic
B. Bandwidth considerations must be taken into account since IDS is deployed inline to traffic flow
C. An IDS analyze a copy of the monitored traffic and not the actual forwarded packet
D. Traffic impact considerations are increased when deploying an IDS over an IPS sensor

Answer: C

Question 9.
In which two locations in an enterprise network can an IPS sensor be placed? (Choose two.)

A. Between two layer devices without trunking
B. Between two Layer 2 devices with trunking
C. Between a Layer 2 device and a Layer 3 device with trunking
D. Bridging two VLANs on one switch
E. Bridging VLANs on two switches

Answer: A, B

Question 10.
DRAG DROP
You work as a network technician at ITCertKeys.com. Your boss, Mrs. ITCertKeys, in Cisco NAC appliance components. Match the applications with the appropriate descriptions. 

Answer:

Explanation:
Network admission control (NAC) is a collection of technologies that can be used to enhance network security services. Specifically, NAC can perform posture validation, which ensures that only permitted devices can communicate on the network. 
NAC appliance has Manager, servers and client like:
1. NAC Manager
2. NAC Server
3. NAC Profile Server
4. NAC Guest Server
5. NAC Client Agent
1. Cisco NAC Appliance Manager (Cisco NAM): Acts as a NAC Appliance administration server for defining policies
2. Cisco NAC Appliance Server (Cisco NAS): Acts as a policy enforcement server between the trusted and untrusted networks
3. Cisco NAC Appliance Agent (Cisco NAA): Acts as an optional agent for Windows-based clients
4. NAC Appliance Policy Updates: Checks the status of updates applied to operating systems, antivirus signatures, and other client software


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Braindumps for "642-642" Exam

Quality of Service (QOS)

 Question 1.
Which of the following markers can be set by Committed Access Rate (CAR)? (Choose all that apply.)

A. DSCP bits
B. QoS Group
C. ATM CLP bit
D. Frame Relay DE bit
E. IP precedence bits

Answer: A, B, E

Explanation:
 
Source: Cisco IP QoS Introduction, Page 64

Question 2.
What makes the DiffServ model more scalable than the IntServ model? (Choose all that apply.)

A. DiffServ makes use of per-aggregate QoS instead of per-flow QoS.
B. DiffServ makes use of hop-by-hop signaling which allows DiffServ to scale to a larger number of application flows.
C. DiffServ is capable of implementing admission control either locally on the routers or be offloaded to a central policy server using the COPS protocol.
D. DiffServ routers are not compelled to track the state information for each individual flow.

Answer: A, D

Incorrect:
B. No hop-by-hop signaling uses per-hop behavior
C. This is a feature of both models Sources: Cisco IP QoS Introduction
http://www.cisco.com/en/US/tech/ ITK5 43/ ITK7 66/technologies_white_paper09186a00800a3e2f.shtml

Question 3.
The newly appointed ITCertKeys trainee technician wants to know what the benefits of using traffic shaping to implement network rate limiting is.What will your reply be?
(Choose all that apply.)

A. Traffic shaping is an effective tool for rate-limiting VoIP traffic.
B. It will not increase packet loss.
C. It will not add to packet transit delays.
D. Traffic shaping can interact with congestion mechanisms embedded in Frame Relay.
E. Traffic shaping can be used on inbound and outbound traffic on a router.

Answer: B, D

Incorrect:
A. Shaping adds variable delay to traffic, possibly causing jitter
C. A shaper typically delays excess traffic using a buffer
E. This is a feature of policing

Explanation:
 
Source: Cisco IP QoS Traffic Shaping and Policing

Question 4.
Study the Exhibit below carefully:
Router# show interfaces hssi 0/0/0 rate-limit
Hssi0/0/0 45Mbps to R1
Input 
matches: all traffic
params: 20000000 bps, 24000 limit, 24000 extended limit
conformed 8 packets, 428 bytes; action: transmit
exceed 0 packets, 0 bytes; action: drop
last packet: 8680ms ago, current burst: 0 bytes
last cleared 00:03:59 ago, conformed 0 bps, exceed 0 bps
Output 
matches: all traffic
params: 20000000 bps, 24000 limit, 24000 extended limit
conformed 0 packets, 0 bytes; action: transmit
exceed 0 packets, 0 bytes; action; drop
last packet: 8680ms ago, current burst: 0 bytes
last cleared 00:03:59 ago, conformed 0 bps, exceed 0 bps
According to the exhibit, the router has been configured with Committed Access Rate (CAR) to rate limit traffic.

What data rate has the traffic been limited to?

A. 192 Kbps
B. 2400 Kbps
C. 4,5 Mbps
D. 20 Mbps
E. 40 Mbps

Answer: D

Explanation:
 
In the configuration example, CAR is applied on the input and output of a customer interface on the provider edge router. Traffic is policed to 256 Kbps on input and output, with some bursting allowed. All exceeding traffic is dropped at the provider edge. The result of the configuration is that traffic to and from the customer is limited to the average rate of approximately 256kbps (256000 in the configuration) with sustained bursts of approximately 32kbps (4kBps or 4000 in the configuration). Initial bursts at line speed can last up to 3 seconds because the token bucket can hold up to 96000 tokens (bytes) which equals 768000 bits (3 x 256000 bits).
Source: Cisco IP QoS Traffic Shaping and Policing, Page 4-91

Question 5:
The newly appointed ITCertKeys trainee technician wants to know which type of software queuing mechanism is Frame Relay Traffic Shaping implemented with on the physical interface. 

What will your reply be?

A. Priority Queuing (PQ)
B. Custom Queuing (CQ)
C. FIFO
D. Weighted Fair Queuing (WFQ)
E. IP Real Time Transport Protocol (RTP) Priority

Answer: D

Explanation:
 
Works on any (sub) interface type  
Shapes traffic on that (sub)interface basis
Can use any physical interface queuing (FIFO, PQ, CQ or WFQ)
Only uses WFQ as the shaping queue (that is, on the input of the shaper)
In contrast, Frame Relay Traffic Shaping:
Works only on Frame Relay (sub) interfaces
Shapes traffic inside individual FR Virtual Circuits
Only permits WFQ as the physical interface queuing method
Can use any queuing method as the shaping queue (that is, on the input of the shaper)
Source: Cisco IP QoS Traffic Shaping and Policing, Page 4-47

Question 6.
Why is it beneficial to make use of Enhanced LMI (ELMI) on Cisco networks?

A. ELMI is responsible for providing virtual provisioning tools at the edge of the Frame Relay network.
B. ELMI permits routers to dynamically download QoS information from Cisco switches for use in traffic shaping or for congestion management purposes.
C. ELMI provides the router the ability to use additional QoS tools including rate limiting with CAR and the Modular QoS Command Line Interface (MQC).
D. ELMI allows the router to deliver packets at the line rate of the Frame Relay interface, regardless of the condition of the Frame Relay network.

Answer: B

Explanation:
 
Source: Cisco IP QoS Traffic Shaping and Policing, Page 4-58

Question 7. 
Which of the following statements are true when you compare DSCP and IP Precedence to each other? (Choose all that apply.)

A. DSCP is backwards compatible with IP Precedence.
B. DSCP cannot be easily mapped into QoS because of its expanded classification options.
C. DSCP is more granular the IP Precedence, since more marking combinations are available.
D. DSCP appears stubby when compared IP Precedence, since devices make use of DSCP as defined in RFC exclusively.
E. DSCP is 6 bits long and IP Precedence is 3 bits long.
F. DSCP is more restrictive than IP Precedence, since devices are only allowed to use DSCP as defined in RFCs.

Answer: A, C, E

Explanation:
A)DSCP is backward compatible with IP Precedence (Class Selector Code point, RFC 1812) but not with the ToS byte definition from RFC 791 ("DTR" bits) Reference: Introduction to IP QoS (Course) p.45

Question 8.
Which MQC command would you use to perform marking properly?

A. precedence 5
B. ip precedence 5
C. set ip precedence 5
D. set ip mark precedence 5
E. mark ip precedence 5

Answer: C

Explanation:
IP precedence is encoded into the three high-order bits of the ToS field in the IP header. It supports eight classes of which two are reserved and should not be used for user-defined classes (IP precedence 6 and 7). IP precedence 0 is the default value and is usually used for the best-effort class. The set ip precedence command marks packets of a class with the specified precedence value. 

Reference: 
Introduction to IP QoS (Course) p.9-104

Question 9.
Which of the following statements aptly describes a network well designed for QoS?

A. Packets are classified at each router, based on as many detail as possible, typically using extended IP ACLs to match the packets for classification.
B. Packets are classified at each router, based on socket address only, typically using extended IP ACLs to match the packets for classification.
C. Packets are classified and marked, close to the edge of the network. The packets are treated differently based on this marking at the routers in the middle of the network.
D. Packets are classified based on different parameters, but close to the edge of the network. The packets are automatically characterized based on flow at the routers.
E. Packets are classified based on socket address, at the router closest to the source of the traffic. The packets are automatically characterized based on flow at the routers.

Answer: C

Explanation:
To achieve the same level of quality in both directions the packets going to and coming from the customer network must first be classified and marked.
Classification and marking packets going to the customer network is a more difficult task because:
1) Classifying and marking must be performed on all edge routers.
2) Classifying and marking requires the identification of the customer network. Using PBR, CAR, CB-Policing or CB-Marking does not scale because it involves the use of access lists (this is especially difficult if customer networks are dynamically learned via BGP).

Reference: 
Introduction to IP QoS (Course) p.2-35

Question 10.
The newly appointed ITCertKeys trainee technician wants to know which bit in the ATM header can be marked by the Class Based Marker to extend IP QoS policy into an ATM network. 

What will your reply be?

A. DE
B. PTI
C. FECN
D. CLP
E. BECN

Answer: D

Explanation:
 
The ATM CLP Setting feature somewhat allows users to extend their IP QoS policies into an ATM network by setting the ATM CLP bit in ATM cells based on the IP Precedence value of the packets being sent. As congestion occurs in the ATM network, cells with the CLP bit set are more likely to be dropped, resulting in improved network performance for high priority traffic and applications. The set atm-clp command marks packets of a class with the ATM CLP bit as a part of an input or output policy.
Source: Cisco IP QoS Modular QoS CLI Service Policy, Page 9-110


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Braindumps for "642-825" Exam

Implementing Secure Converged Wide Area Networks

 Question 1.
A few small ITCertKeys locations use HFC cable to connect to the ITCertKeys WAN.

Which HFC cable network statement is true about the downstream data channel to the customer and the upstream data channel to the service provider?

A. The upstream data path is assigned a channel in a higher frequency range than the downstream path has.
B. The downstream data path is assigned a 30 MHz channel and the upstream data path is assigned a 1 MHz channel.
C. The downstream data path is assigned a fixed bandwidth channel and the upstream data path uses a variable bandwidth channel.
D. Both upstream and downstream data paths are assigned in 6 MHz channels.
E. None of the above.

Answer: D

Explanation:
Hybrid fiber-coaxial (HFC): A mixed optical-coaxial network in which optical fiber replaces some or all of the traditional trunk portion of the cable network. The HFC architecture is the evolution of an initial cable system and signifies a network that incorporates both optical fiber along with coaxial cable to create a broadband network. By upgrading a cable plant to an HFC architecture, you can deploy a data network over an HFC system to offer high-speed Internet services and you can serve more subscribers. The cable network is segmented into smaller service areas in which fewer amplifiers are cascaded after each optical node-typically five or fewer. The tree-and-branch network architecture for HFC can be a fiber backbone, cable area network, superdistribution, fiber to the feeder, or a ring. 

Downstream: An RF signal transmission (TV channels, data) from source (headend) to the destination (subscribers). Downstream is also called a forward path.
Upstream: An RF signal transmission opposite to downstream-from subscribers to the headend. Upstream is also called a return or reverse path.
Delivering services over a cable network requires different RF frequencies-the outgoing frequencies are in the 50-to-860 MHz range, the incoming are in the 5-to-42 MHz range. To deliver data services over a cable network TV channels which usually operate at 6 MHz range for the downstream, and 6 MHz or less (for asymmetric cable connections) for upstream traffic from the corresponding frequency range are usually used.

Question 2.
Many small ITCertKeys branch offices use broadband cable for data connection access. 

Which three modulation signaling standards are used in broadband cable technology? (Select three)

A. S-Video
B. NTSC
C. SECAM
D. PAL
E. FEC
F. FDM
G. MLP

Answer: B, C, D

Explanation:
Broadband: Data transmission where multiple pieces of data are sent simultaneously to increase the effective rate of transmission. In cable systems, the term broadband refers to the frequency-division multiplexing (FDM) of many signals in a wide radio frequency (RF) bandwidth over an HFC network, and the capability to handle vast amounts of information. NTSC is a North American TV technical standard for analog TV systems. The standard was created in 1941 and is named after the National Television System Committee formed in 1940. The standard uses a 6-MHz modulated signal. PAL is a color encoding system used in broadcast television systems in most of Europe, Asia, Africa, Australia, Brazil, and Argentina, and uses a 6-MHz, 7-MHz, or 8-MHz modulated signal. The color difference signals an alternate phase at the horizontal line rate. SECAM is an analog color TV system used in France and certain Eastern European countries that uses an 8-MHz modulated signal.

Question 3.
Some of the smaller ITCertKeys locations use HFC cable to connect to the ITCertKeys WAN. 

Which two statements are true about broadband cable (HFC) systems? (Select two)

A. Cable modems operate at Layers 1, 2, and 3 of the OSI model.
B. Cable modems operate at Layers 1 and 2 of the OSI model.
C. A function of the cable modem termination system is to convert the digital data stream from the end user host into a modulated RF signal for transmission onto the cable system.
D. Cable modems only operate at Layer 1 of the OSI model.
E. A function of the cable modem termination system (CMTS) is to convert the modulated signal from the cable modem into a digital signal.

Answer: B, E

Explanation:
Hybrid fiber-coaxial (HFC): A mixed optical-coaxial network in which optical fiber replaces some or all of the traditional trunk portion of the cable network. The HFC architecture is the evolution of an initial cable system and signifies a network that incorporates both optical fiber along with coaxial cable to create a broadband network. By upgrading a cable plant to an HFC architecture, you can deploy a data network over an HFC system to offer high-speed Internet services and you can serve more subscribers. The cable network is segmented into smaller service areas in which fewer amplifiers are cascaded after each optical node-typically five or fewer. The tree-and-branch network architecture for HFC can be a fiber backbone, cable area network, superdistribution, fiber to the feeder, or a ring.

Question 4.
A ITCertKeys remote user is getting Internet access from the local cable provider. 

When an individual is connected to the Internet by way of a CATV cable service, what kind of traffic is considered upstream traffic?

A. Traffic going from the user's home traveling to the headend.
B. Broadcast traffic, including the cable TV signals.
C. Traffic between the headend and the TV signal.
D. Traffic between the headend and the supplier antenna.
E. Traffic from outside the local cable segment serving the user's home.
F. All of the above can be considered upstream

Answer: A

Explanation:
In the CATV space, the downstream channels in a cable plant (cable head-end to subscribers) is a point-to-multipoint channel. This does have very similar characteristics to transmitting over an Ethernet segment where one transmitter is being listened to by many receivers. The major difference is that base-band modulation has been replaced by a more densely modulated RF carrier with very sophisticated adaptive signal processing and forward error correction (FEC). In the upstream direction (subscriber cable modems transmitting towards the head-end) the environment is many transmitters and one receiver. This introduces the need for precise scheduling of packet transmissions to achieve high utilization and precise power control so as to not overdrive the receiver or other amplifier electronics in the cable system. Since the upstream direction is like a single receiver with many antennas, the channels are much more susceptible to interfering noise products. In the cable industry, we generally call this ingress noise. As ingress noise is an inherent part of CATV plants, the observable impact is an unfortunate rise in the average noise floor in the upstream channel. To overcome this noise jungle, upstream modulation is not as dense as in the downstream and we have to use more effective FEC as used in the downstream. 

Reference: 
http://www.cisco.com/warp/public/759/ipj_1-3/ipj_1-3_catv.html

Question 5.
A new cable modem was shipped to the home of a ITCertKeys user, where it is being installed for the first time. 

When a DOCSIS 1.1 compliant cable modem first initializes, (boots up) what does it do?

A. Establishes IP connectivity (DHCP).
B. Determines the time of day.
C. Requests a DOCSIS configuration file from a TFTP server.
D. Scan for a downstream channel and the establishment of timing synchronization with the CMTS.
E. None of the above.

Answer: D

Explanation:
According to the DOCSIS (Data-over-Cable Service Interface Specifications) when you first power up a cable modem it starts scanning (starting at a low frequency) for a cable signal. When it 'hears' a cable modem stream it listens for a broadcast (from the service provider) which contains information (ie. frequency) needed to talk back with the head end. It then 'talks back' and if it communicates the right authentication information, it is allowed to proceed.

References: 
Page 225 of the CCNP Self-Study BCRAN (642-821) ISBN: 1-58720-084-8
http://www.cisco.com/en/US/products/hw/cable/ps2217/products_feature_guide_chapter09186a008019b57f.htm

Question 6.
You are building a small network at your home and you intend on connecting your cable modem to a Cisco router.

Which router interface would you connect the modem to?

A. Synchronous serial
B. Asynchronous serial
C. Ethernet
D. auxiliary
E. BRI

Answer: C

Explanation:
In certain environments where a non Cisco Cable Modem (CM) is used, and the CM is only capable of bridging, a Cisco router such as the Cisco 806 can be connected to the Cable Modem via the Ethernet interface. The routing can then be performed by the Cisco router behind the Cable Modem and the Client PC or Customer Premises Equipment (CPE) will be connected to the Cisco router. Network Address Translation (NAT) can  then be configured on the Cisco router.

When the Cisco router is connected behind the Cable Modem the first problem that might be encountered is not obtaining an IP address dynamically on the Cisco router's Ethernet interface. Most Internet Service Providers (ISPs) allow only one host or PC behind the Cable Modem. Some ISPs assign an IP address to the PC based on the host name. Therefore, if you have a Cisco router behind the Cable Modem, then the host name for the router configured using the hostname command should be the same host name given by the ISP.
Example:
 
Question 7.
When a cable modem is being provisioned to operate with a host system for Internet services, which two options must occur before Layer 1 and 2 connectivity can occur? (Choose two)

A. The cable modem must request an IP address and core configuration information from a Dynamic Host Configuration Protocol (DHCP) server.
B. The cable modem powering up must scan and lock on the RF data channel in the downstream path.
C. The modem must request a DOCSIS configuration file from a TFTP server.
D. The cable modem must register with the CMTS.
E. The modem must read specific maintenance messages in the downstream path.

Answer: B, E

Explanation:
According to the DOCSIS (Data-over-Cable Service Interface Specifications) when you first power up a cable modem it starts scanning (starting at a low frequency) for a cable signal. When it 'hears' a cable modem stream it listens for a broadcast (from the service provider) which contains information (ie. frequency) needed to talk back with the head end. It then 'talks back' and if it communicates the right authentication information, it is allowed to proceed. Once these steps are completed, layers 1 and 2 will be operational.

Question 8.
How is cable broadband technology able to transmit downstream and upstream data while at the same time delivering television content?

A. The cable operator uses the VHF hyperband to transmit and receive data signals.
B. The cable operator assigns any available spectrum to data, depending on how its own television spectrum is being used.
C. The cable operator uses specific bandwidths for data signals specified by DOCSIS.
D. The cable operator places its data signals into clean areas where there is no interference from noise or other signals.

Answer: C

Explanation:
Developed by CableLabs and approved by the ITU in March 1998, Data Over Cable Service Interface Specification (DOCSIS) defines interface standards for cable modems and supporting equipment. In a cable TV system, signals from the various channels are each given a 6-MHz slice of the cable's available bandwidth and then sent down the cable to your house. In some systems, coaxial cable is the only medium used for distributing signals.
 
When a cable company offers Internet access over the cable, Internet information can use the same cables because the cable modem system puts downstream data -- data sent from the Internet to an individual computer -- into a 6-MHz channel. On the cable, the data looks just like a TV channel. So Internet downstream data takes up the same amount of cable space as any single channel of programming. Upstream data -- information sent from an individual back to the Internet -- requires even less of the cable's bandwidth, just 2 MHz, since the assumption is that most people download far more information than they upload.

Question 9.
ITCertKeys operates a DSL network. What does the "dsl operating-mode auto" command configure on a Cisco router?

A. It configures a Cisco router to automatically detect the proper modulation method to use when connecting an ATM interface.
B. It configures a Cisco router to automatically detect the proper DSL type (ADSL, IDSL, HDSL, VDSL) to use when connecting an ATM interface.
C. It configures a Cisco router to automatically detect the proper encapsulation method to use when connecting an ATM interface.
D. It configures a Cisco router to automatically detect the proper authentication method to use when connecting an ATM interface.
E. None of the above

Answer: A

Explanation:
dsloperating-mode auto interface configuration command to specify that the router automatically detect the DSL modulation that the service provider is using and set the DSL modulation to match. An incompatible DSL modulation configuration can result in failure to establish a DSL connection to the DSLAM of the service provider
Example:
 
Question 10.
ITCertKeys is a DSL service provider using providing xDSL to its customers. 

Which statement about xDSL implementations is true?

A. All xDSL standards operate in lower frequencies than the POTS system and can therefore coexist on the same media.
B. Other than providing higher data rates, HDSL is identical to ADSL.
C. The ADSL standard operates in higher frequencies than the POTS system and can therefore coexist on the same media.
D. The HDSL standard operates in higher frequencies than the POTS system and can therefore coexist on the same media.
E. All xDSL standards operate in higher frequencies than the POTS system and therefore can coexist on the same media.
F. None of the above.

Answer: C

Explanation:
DSL is not a complete end-to-end solution, but rather a physical layer transmission technology similar to dial, cable, or wireless. DSL connections are deployed in the "last mile" of a local telephone network-the local loop. The connection is set up between a pair of modems on either end of a copper wire extending between the customer premises equipment (CPE) and the DSL access multiplexer (DSLAM). A DSLAM is the device located at the central office (CO) of the provider and concentrates connections from multiple DSL subscribers. The term xDSL covers a number of DSL variations, such as ADSL, high-data-rate DSL (HDSL), Rate Adaptive DSL (RADSL), symmetric DSL (SDSL), ISDN DSL (IDSL), and very-high-data-rate DSL (VDSL). DSL types not using the voice frequencies band allow DSL lines to carry both data and voice signals simultaneously (for example, ADSL and VDSL), while other DSL types occupying the complete frequency range can carry data only (for example, SDSL and IDSL). Data service provided by a DSL connection is always-on. The data rate that DSL service can provide depends upon the distance between the subscriber and the CO. The smaller the distance, the higher data rate can be achieved. If close enough to a CO offering DSL service, the subscriber might be able to receive data at rates of up to 6.1 Mbps out of a theoretical 8.448 Mbps maximum.


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Optimizing Converged Cisco Networks

 Question 1.
You need to implement QoS for the ITCertKeys VOIP network. 

Which three statements are true about the data traffic characteristics of voice traffic? (Select three)

A. Voice packets require TCP for rapid retransmission of dropped packets.
B. Latency is not a concern as long as jitter is kept below 30 ms.
C. Voice packets require a fairly constant bandwidth reserved for voice control traffic as well as 
    for the voice payload.
D. Voice packets do not require a specific type of queuing.
E. Latency must be kept below 150 ms.
F. Voice packets are rather small

Answer: C, E, F

Explanation:
QoS refers to the ability of a network to provide improved service to selected network traffic over various underlying technologies including Frame Relay, ATM, Ethernet and 802.3 networks, SONET, and IP-routed networks.

QoS features provide improved and more predictable network service by offering the following services:
1. Dedicated bandwidth
2. Improved loss characteristics
3. Congestion management and avoidance
4. Traffic shaping
5. Prioritization of traffic
Voice quality is directly affected by all three QoS quality factors such as loss, delay, and delay variation. Loss causes voice clipping and skips. Industry standard codec algorithms can correct for up to 30 ms of lost voice. Cisco Voice over IP (VoIP) technology uses 20 ms samples of voice payload per VoIP packet. Only a single Real Time Transport (RTP) packet could be lost at any given time. If two successive voice packets are lost, the 30 ms correctable window is exceeded and voice quality begins to degrade.

Delay can cause voice quality degradation if it is above 200 ms. If the end-to-end voice delay becomes too long, the conversation sounds as if two parties are talking over a satellite link or a CB radio. The ITU standard for VoIP, G.114, states that a 150 ms one-way delay budget is acceptable for high voice quality. With respect to delay variation, there are adaptive jitter buffers within IP Telephony devices. These buffers can usually compensate for 20 to 50 ms of jitter.

Question 2.
ITCertKeys uses G.711 for the VOIP calls. When analog signals are digitized using the G.711 codec, voice samples are encapsulated into protocol data units (PDUs) involving which three headers? (Select three)

A. UDP
B. RTP
C. IP
D. TCP
E. Compressed RTP
F. H.323

Answer: A, B, C

Explanation:
When a VoIP device, such as a gateway, sends voice over an IP network, the digitized voice has to be encapsulated into an IP packet. Voice transmission requires features not provided by the IP protocol header; therefore, additional transport protocols have to be used. Transport protocols that include features required for voice transmission are TCP, UDP, and RTP. VoIP utilizes a combination of UDP and RTP.

Question 3.
VOIP has been rolled out to every ITCertKeys location.

What are three features and functions of voice (VOIP) traffic on a network? (Select three)

A. Voice traffic is bursty
B. Voice traffic is retransmittable
C. Voice traffic is time-sensitive
D. Voice traffic is bandwidth intensive
E. Voice traffic is constant
F. Voice traffic uses small packet sizes

Answer: C, E, F

Explanation:
The benefits of packet telephony networks include 
i. More efficient use of bandwidth and equipment: Traditional telephony networks use a 64-kbps channel for every voice call. Packet telephony shares bandwidth among multiple logical connections.
ii. Lower transmission costs: A substantial amount of equipment is needed to combine 64-kbps channels into high-speed links for transport across the network. Packet telephony statistically multiplexes voice traffic alongside data traffic. This consolidation provides substantial savings on capital equipment and operations costs.
iii. Consolidated network expenses: Instead of operating separate networks for voice and data, voice networks are converted to use the packet-switched architecture to create a single integrated communications network with a common switching and transmission system. The benefit is significant cost savings on network equipment and operations.
iv. Improved employee productivity through features provided by IP telephony: IP phones are not only phones, they are complete business communication devices. They offer directory lookups and access to databases through Extensible Markup Language (XML) applications. These applications allow simple integration of telephony into any business application. For instance, employees can use the phone to look up information about a customer who called in, search for inventory information, and enter orders. The employee can be notified of a issue (for example, a change of the shipment date), and with a single click can call the customer about the change. In addition, software-based phones or wireless phones offer mobility to the phone user.

Question 4.
ITCertKeys is rolling out an H.323 VOIP network using Cisco devices. 

Which IOS feature provides dial plan scalability and bandwidth management for H.323 VoIP implementations?

A. Digital Signal Processors
B. Call Routing
C. Gatekeeper
D. Call Admission Control
E. None of the above

Answer: C

Explanation:
Enterprise voice implementations use components such as gateways, gatekeepers, Cisco Unified CallManager, and IP phones. Cisco Unified CallManager offers PBX-like features to IP phones. Gateways interconnect traditional telephony systems, such as analog or digital phones, PBXs, or the public switched telephone network (PSTN) to the IP telephony solution. Gatekeepers can be used for scalability of dial plans and for bandwidth management when using the H.323 protocol.

Question 5.
A Cisco router is being used as a VOIP gateway to convert voice signals in the ITCertKeys network. 

What steps are taken when a router converts a voice signal from analog to digital form? (Select two)

A. Quantization
B. Serialization
C. Packetization
D. Sampling

Answer: A, D

Explanation:
Step 1 Sampling: The analog signal is sampled periodically. The output of the sampling is a pulse amplitude modulation (PAM) signal.
Step 2 Quantization: The PAM signal is matched to a segmented scale. This scale measures the amplitude (height) of the PAM signal.
Step 3 Encoding: The matched scale value is represented in binary format.
Step 4 Compression: Optionally, voice samples can be compressed to reduce bandwidth requirements. Analog-to-digital conversion is done by digital signal processors (DSPs), which are located on the voice interface cards. The conversion is needed for calls received on analog lines, which are then sent out to a packet network or to a digital voice interface.

Question 6.
You need to implement the proper IOS tools to ensure that VOIP works over the ITCertKeys network. 

Which queuing and compression mechanisms are needed to effectively use the available bandwidth for voice traffic? (Select two)

A. Priority Queuing (PQ) or Custom Queuing (CQ)
B. Real-Time Transport Protocol (RTP) header compression
C. Low Latency Queuing (LLQ)
D. Class-Based Weighted Fair Queuing (CBWFQ)
E. TCP header compression
F. UDP header compression

Answer: D, E

Explanation:
1. Class-based weighted fair queuing (CBWFQ) extends the standard WFQ functionality to provide support for user-defined traffic classes. By using CBWFQ, network managers can define traffic classes based on several match criteria, including protocols, access control lists (ACLs), and input interfaces. A FIFO queue is reserved for each class, and traffic belonging to a class is directed to the queue for that class. More than one IP flow, or "conversation", can belong to a class. Once a class has been defined according to its match criteria, the characteristics can be assigned to the class. To characterize a class, assign the bandwidth and maximum packet limit.  The bandwidth assigned to a class is the guaranteed bandwidth given to the class during congestion. CBWFQ assigns a weight to each configured class instead of each flow. This weight is proportional to the bandwidth configured for each class. Weight is equal to the interface bandwidth divided by the class bandwidth. Therefore, a class with a higher bandwidth value will have a lower weight. By default, the total amount of bandwidth allocated for all classes must not exceed 75 percent of the available bandwidth on the interface. The other 25 percent is used for control and routing traffic. The queue limit must also be specified for the class. The specification is the maximum number of packets allowed to accumulate in the queue for the class. Packets belonging to a class are subject to the bandwidth and queue limits that are configured for the class.
2. TCP/IP header compression subscribes to the Van Jacobson Algorithm defined in RFC 1144. TCP/IP header compression lowers the overhead generated by the disproportionately large TCP/IP headers as they are transmitted across the WAN. TCP/IP header compression is protocol-specific and only compresses the TCP/IP header. The Layer 2 header is still intact and a packet with a compressed TCP/IP header can still travel across a WAN link. TCP/IP header compression is beneficial on small packets with few bytes of data such as Telnet. Cisco's header compression supports Frame Relay and dial-on-demand WAN link protocols. Because of processing overhead, header compression is generally used at lower speeds, such as 64 kbps links.

Question 7.
You want to ensure the highest call quality possible for all VOIP calls in the ITCertKeys network. 

Which codec standard would provide the highest voice-quality, mean opinion score (MOS)?

A. G.711, PCM
B. G.729, CS-ACELP
C. G.729A, CS-ACELP
D. G.728, LDCELP
E. None of the above

Answer: A

Explanation:
When a call is placed between two phones, the call setup stage occurs first. As a result of this process, the call is logically set up, but no dedicated circuits (lines) are associated with the call. The gateway then converts the received analog signals into digital format using a codec, such as G.711 or G.729 if voice compression is being used. When analog signals are digitized using the G.711 codec, 20 ms of voice consists of 160 samples, 8 bits each. The result is 160 bytes of voice information. These G.711 samples  (160 bytes) are encapsulated into an RTP header (12 bytes), a UDP header (8 bytes), and an IP header (20 bytes). Therefore, the whole IP packet carrying UDP, RTP, and the voice payload has a size of 200 bytes. When G.711 is being used, the ratio of header to payload is smaller because of the larger voice payload. Forty bytes of headers are added to 160 bytes of payload, so one-fourth of the G.711 codec bandwidth (64 kbps) has to be added. Without Layer 2 overhead, a G.711 call requires 80 kbps.

Question 8.
When a router converts analog signals to digital signals as part of the VoIP process, it performs four separate steps. 

From the options shown below, which set of steps contains the steps in their correct sequence?

A. encoding
    quantization
    optional compression
    sampling
B. optional compression
    encoding
    sampling
    quantization
C. sampling
    quantization
    encoding
    optional compression
D. optional compression
    sampling
    encoding
    quantization
E. sampling
    quantization
    optional compression
    encoding
F. encoding
    optional compression
    quantization
    sampling
G. None of the above

Answer: C

Explanation:
Step 1 Sampling: The analog signal is sampled periodically. The output of the sampling is a pulse amplitude modulation (PAM) signal.
Step 2 Quantization: The PAM signal is matched to a segmented scale. This scale measures the amplitude (height) of the PAM signal.
Step 3 Encoding: The matched scale value is represented in binary format.
Step 4 Compression:
Optionally, voice samples can be compressed to reduce bandwidth requirements. Analog-to-digital conversion is done by digital signal processors (DSPs), which are located on the voice interface cards. The conversion is needed for calls received on analog lines, which are then sent out to a packet network or to a digital voice interface.

Question 9.
ITCertKeys has determined that during its busiest hours, the average number of internal VoIP calls across the WAN link is four (4). Since this is an average, the WAN link has been sized for six (6) calls with no call admission control. 

What will happen when a seventh call is attempted across the WAN link?

A. The seventh call is routed via the PSTN.
B. The call is completed, but all calls have quality issues.
C. The call is completed but the seventh call has quality issues.
D. The call is denied and the original six (6) calls remain.
E. The call is completed and the first call is dropped.
F. None of the above.

Answer: B

Explanation:
IP telephony solutions offer Call Admission Control (CAC), a feature that artificially limits the number of concurrent voice calls to prevent oversubscription of WAN resources. Without CAC, if too many calls are active and too much voice traffic is sent, delays and packet drops occur. Even giving Real-Time Transport Protocol (RTP) packets absolute priority over all other traffic does not help when the physical bandwidth is not sufficient to carry all voice packets. Quality of service (QoS) mechanisms do not associate individual RTP packets with individual calls; therefore, all RTP packets are treated equally. All RTP packets will experience delays, and any RTP packets may be dropped. The effect of this behavior is that all voice calls experience voice quality degradation when oversubscription occurs. It is a common misconception that only calls that are beyond the bandwidth limit will suffer from quality degradation. CAC is the only method that prevents general voice quality degradation caused by too many concurrent active calls.

Question 10.
While planning the new ITCertKeys VOIP network, you need to determine the size of the WAN links to use. To do this, you need to calculate the bandwidth required by each call.

Which three pieces of information are used to calculate the total bandwidth of a VoIP call? (Select three)

A. The serialization of the interface
B. The quantization
C. The TCP overhead
D. The packetization size
E. The UDP overhead
F. The packet rate

Answer: D, E, F

Explanation:
Packet rate: Packet rate specifies the number of packets sent in a certain time interval. The packet rate is usually specified in packets per second (pps). Packet rate is the multiplicative inverse of the packetization period. The packetization period is the amount of voice (time) that will be encapsulated per packet, and is usually specified in milliseconds. Packetization size: Packetization size specifies the number of bytes that are needed to represent the voice information that will be encapsulated per packet. Packetization size depends on the packetization period and the bandwidth of the codec used. IP overhead: IP overhead specifies the number of bytes added to the voice information during IP encapsulation. When voice is encapsulated into Real-Time Transport Protocol (RTP), User Datagram Protocol (UDP), and IP, the IP overhead is the sum of all these headers.

Data link overhead: Data-link overhead specifies the number of bytes added during data-link encapsulation. The data-link overhead depends on the used data-link protocol, which can be different per link. Tunneling overhead: Tunneling overhead specifies the number of bytes added by any security or tunneling protocol, such as 802.1Q tunneling, IPsec, Generic Route Encapsulation (GRE), or Multiprotocol Label Switching (MPLS). This overhead must be considered on all links between the tunnel source and the tunnel destination.



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Cisco Lifecycle Services Express

 Question 1.
Which service component within the design phase does the following describe? Consider the customer's defined business requirements and the associated performance, availability, resiliency, maintainability, resource capacity, and security criteria used to measure and assure the delivery of the required services.

A. Staging Plan
B. Implementation Plan
C. Business Plan
D. Detailed Design Validation
E. Project Kick-off

Answer: B

Question 2.
Which service component within the prepare phase validates the features and functionality documented in the high-level design of a solution?

A. Proof of Concept
B. High-level Design Development
C. Business Case Development
D. Business Requirements Development
E. Operations Technology Strategy Development
F. Technology Strategy Development

Answer: A

Question 3.
In the implement phase, project close-out involves which of the following tasks?

A. deliver education based on Staff Planning Development reports
B. conduct Engagement Profitability Assessment
C. execute network migration plan
D. customize ongoing support hand-off kit

Answer: B

Question 4.
In the implement phase, project close-out involves which of the following tasks?

A. deliver education based on Staff Planning Development reports
B. customize ongoing support hand-off kit
C. conduct Engagement Profitability Assessment
D. execute network migration plan

Answer: C

Question 5.
In the design phase, performing a needs analysis to identify the resource requirements and skills needed to deliver and support a network infrastructure is an activity of which service component?

A. Implementation Kick-off Meeting
B. Implementation Plan
C. Project Kick-off
D. Staging Plan
E. Migration Plan
F. Staff Plan Development

Answer: F

Question 6.
In the implement phase, solution implementation involves which of the following task?

A. Execute test cases
B. Configure core products
C. Perform preliminary Site Assessment
D. Analyze high-level design and technology requirements of the customer

Answer: B

Question 7.
Which one of the following describes the task for completing an operations readiness assessment in the plan phase?

A. determining network elements that will be tested
B. reviewing hardware and software configuration staging plan
C. entering documentation into knowledge management system
D. developing backup/recovery plan
E. creating an escalation plan
F. obtaining and reviewing operation procedures and policies

Answer: F

Question 8.
In the implement phase, solution implementation involves which of the following task? 

A. perform preliminary Site Assessment
B. configure core products
C. execute test cases
D. analyze high-level design and technology requirements of the customer

Answer: B

Question 9.
Which of the following definitions best describes services stack within the Cisco Lifecycle Services approach?

A. the minimum set of services that a customer needs to successfully deploy and operate a Cisco technology or solution
B. the required set of services needed to successfully deploy and support Cisco Advanced Technologies
C. the identification of Advanced Technologies to best support business requirements and objectives
D. the minimum set of services required to operate and optimize Cisco Advanced Technologies

Answer: A

Question 10.
Which of the following definitions best describes business requirements development within the prepare phase?

A. a set of service component activities that assesses and documents the business requirements of a customer
B. part of the systems design activity that identifies and documents business requirements to help deploy network technologies
C. part of the high-level design activity that addresses business and technical requirements of the customer
D. a service component activity that analyzes business requirements and recommends the appropriate technology strategy

Answer: A


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